[asterisk-dev] Re: SIPP Testing

Abdul Lateef Khan abdulzu at hotmail.com
Wed Jun 7 22:42:24 MST 2006


Hello,

Can u post me xml file. i was trying with the following commond but it is 
not working.

./sipp -sn uac -d 500 -s 123456 asteriskserver:1221 -l 300 -mp 1221 -i 
asteriskserver:1221

Error:

Resolving asteriskserver...
media IP addr= asteriskserver
2006-06-08 01:41:35: Unable to bind RTP socket.

Could u please help to solve this issue?






>From: "Dan Austin" <Dan_Austin at Phoenix.com>
>Reply-To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Subject: RE: [asterisk-dev] Re: SIPP Testing
>Date: Wed, 7 Jun 2006 22:10:37 -0700
>
>The patch you seek is located in Mantis bugid 5374...
>
>Perhaps it would help.  What is the hardware you are
>Testing against?  If I read the chart correctly you could
>only setup about 3 calls per second which just doesn't
>seem right...
>
>-----Original Message-----
>From: asterisk-dev-bounces at lists.digium.com
>[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Greg
>Boehnlein
>Sent: Wednesday, June 07, 2006 9:48 PM
>To: asterisk-dev at lists.digium.com
>Subject: [asterisk-dev] Re: SIPP Testing
>
>On Wed, 7 Jun 2006, Greg Boehnlein wrote:
>
> > Hello,
> > 	I am working on a bunch of issues, one of which is trying to
>track
> > down proof that using CONFIG_ZAPTEL_MMX w/ a Centos 4.3 2.6.9-34.0.1
> > kernel causes FPU errors w/ various codecs. This requires me to have a
>
> > high volume of SIP calls doing lots of Ulaw to G729 transcoding over a
>
> > period of time, using a modified g729 binary from Digium that logs
>when
> > the errors occur.
> > 	In doing this testing, I have setup a box w/ the latest SIPP
> > traffic generator (http://sipp.sourceforge.net) to make boatloads of
>120
> > second calls to a Music On Hold extension that simply plays Ulaw based
>
> > MOH to the caller.
>
>Just wanted to follow up on my post, archive it for the world and
>provide
>some insight into what the issue was.
>
>Asterisk's 1.2 RTP stack requires bidirectional traffic to actually send
>
>traffic back. There was a patch in Mantis a while back that did
>asynchronous RTP, but I was unable to find it. However, the fact that
>you
>need to receive an RTP packet to SEND an RTP packet is the reason why no
>
>RTP was originating from the Asterisk server.
>
>So.. here is what I did to get things working:
>
>1. Installed libpcap and libnet on my development box.
>2. Built SIPP w/ Pcap Play suppor using "make pcapplay"
>3. I grabbed a g729 RTP strean from a SNOM phone w/ TCPDUMP:
>
>tcpdump -T rtp -vvv dst 207.166.192.106 -w g729.pcap
>
>4. I then dumped the "uac_pcap" XML file and made some tweaks:
>
>./sipp -sd uac_pcap > uac_pcap.xml
>
>My edits consisted of modifying the following line:
>
><exec play_pcap="pcap/g711a.pcap"/>
>
>To
>
><exec play_pcap="pcap/g729.pcap"/>
>
>And, I changed:
>
>a=rtpmap:0 PCMU/8000
>
>To
>
>a=rtpmap:18 G729/8000
>
>5. I then executed the following:
>
>./sipp -sf uac-g729.xml -d 10000 -s 451 gw4.n2net.net -l 96 -mi
>207.166.192.254 -mp 5606 -i 207.166.192.254
>
>And presto.. LOTS of calls and LOTS of RTP traffic. Once you send one
>RTP
>packet to the Asterisk server, it responds back to the sipp RTP mirror
>port, and things just chug along after that..
>
>My results (not to promising)
>
>------------------------------ Scenario Screen -------- [1-4]: Change
>Screen --
>   Call-rate(length)     Port   Total-time  Total-calls  Remote-host
>10.0(10000 ms)/1.000s   5060     877.47 s         2485
>207.166.192.184:5060(UDP)
>
>   0 new calls during 1.000 s period      2 ms scheduler resolution
>   2 concurrent calls (limit 96)          Peak was 96 calls, after 9 s
>   1501 out-of-call msg (discarded)
>   1 open sockets
>   6390121 Total RTP pckts                236.576 last period RTP rate
>(kB/s)
>
>                                  Messages  Retrans   Timeout
>Unexpected-Msg
>       INVITE ---------->         2485      11980     2301
>          100 <----------         144       0                   45
>          180 <----------         0         0                   0
>          183 <----------         0         0                   0
>          200 <---------- E-RTD   139       674                 0
>          ACK ---------->         139       674
>               [ NOP ]
>        Pause [   8000ms]         139                           0
>    Var Pause [  10000ms]         139                           0
>          BYE ---------->         139       1209      130
>          200 <----------         6         0                   1
>
>------- Waiting for active calls to end. Press [Ctrl-c] to force exit.
>--------
>
>------------------------------ Scenario Screen -------- [1-4]: Change
>Screen --
>   Call-rate(length)     Port   Total-time  Total-calls  Remote-host
>10.0(10000 ms)/1.000s   5060     878.45 s         2485
>207.166.192.184:5060(UDP)
>
>   0 new calls during 0.986 s period      1 ms scheduler resolution
>   0 concurrent calls (limit 96)          Peak was 96 calls, after 9 s
>   1501 out-of-call msg (discarded)
>   1 open sockets
>   6397343 Total RTP pckts                234.380 last period RTP rate
>(kB/s)
>
>                                  Messages  Retrans   Timeout
>Unexpected-Msg
>       INVITE ---------->         2485      11980     2301
>          100 <----------         144       0                   45
>          180 <----------         0         0                   0
>          183 <----------         0         0                   0
>          200 <---------- E-RTD   139       674                 0
>          ACK ---------->         139       674
>               [ NOP ]
>        Pause [   8000ms]         139                           0
>    Var Pause [  10000ms]         139                           0
>          BYE ---------->         139       1209      132
>          200 <----------         6         0                   1
>
>------------------------------ Test Terminated
>--------------------------------
>
>
>----------------------------- Statistics Screen ------- [1-4]: Change
>Screen --
>   Start Time             | 2006-06-08 00:21:38
>
>   Last Reset Time        | 2006-06-08 00:36:15
>
>   Current Time           | 2006-06-08 00:36:16
>
>-------------------------+---------------------------+------------------
>--------
>   Counter Name           | Periodic value            | Cumulative value
>-------------------------+---------------------------+------------------
>--------
>   Elapsed Time           | 00:00:00:986              | 00:14:38:484
>
>   Call Rate              |    0.000 cps              |    2.829 cps
>
>-------------------------+---------------------------+------------------
>--------
>   Incoming call created  |        0                  |        0
>
>   OutGoing call created  |        0                  |     2485
>
>   Total Call created     |                           |     2485
>
>   Current Call           |        0                  |
>
>-------------------------+---------------------------+------------------
>--------
>   Successful call        |        0                  |        6
>
>   Failed call            |        2                  |     2479
>
>-------------------------+---------------------------+------------------
>--------
>   Response Time          | 00:00:00:000              | 00:00:10:058
>
>   Call Length            | 00:00:50:108              | 00:00:32:841
>
>------------------------------ Test Terminated
>--------------------------------
>
>2006-06-08 00:36:16: Aborting call on UDP retransmission timeout for
>Call-ID '2429-17803 at 207.166.192.254'.
>sipp: There were more errors, enable -trace_err to log them.
>
>[root at dmx-64 sipp.cumulus.2006-03-20]#
>
>--
>     Vice President of N2Net, a New Age Consulting Service, Inc. Company
>          http://www.n2net.net Where everything clicks into place!
>                              KP-216-121-ST
>
>
>
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